The current version of the standardized GPRS system contains four mutually alternative channel coding schemes, which are known as CS-1, CS-2, CS-3 and CS-4. Error correction is based on retransmissions, meaning that the receiving device must acknowledge successfully received packets. Retransmission of unacknowledged packets is attempted until the reception is successful or the packets must be discarded due to a time limit. The decision of which channel coding scheme should be used in each communication connection is on the responsibility of the Packet Control Unit or PCU which is typically a part of the Base Station Subsystem or BSS; more particularly a PCU typically operates at the Base Station Controller or BSC, at the Base Transceiver Station or BTS (also known shortly as the Base Station or BS) or even at a Serving GPRS Support Node or SGSN. The channel coding scheme of a certain communication connection can be dynamically changed according to need. For example in a connection where frequent retransmissions of unsuccessfully received packets are observed it may be worthwhile to introduce a stronger channel coding scheme. Increasing the amount of channel coding lowers the throughput of actual data per packet, so when the connection quality is otherwise good it is advantageous to keep the amount of channel coding fairly low. All said GPRS channel coding schemes are associated with interleaving over a RLC block period (Radio Link Control) which is equal to the duration of four consecutive transmission frames.
The problem of the GPRS radio interface, as well as of many other radio interfaces for packet data communications, viewed in the context of the present invention, is that they are optimized for the transmission of data, meaning particularly the non-real time transmission of files and messages. This is understandable as such, because the packet-switched radio transmission systems have been regarded as logical extensions of wired networks for packet-switched communication between computers. Delay-critical applications such as the real-time transmission of speech and video have had their own circuit-switched transmission systems. Recently, however, applications have arisen that use the packet data networks for the real-time transmission of speech and even images. An example is the technology of Internet calls, where an audio and/or video telephone call is conducted through the Internet.
If we attempt to use the known GPRS radio interface or the known UMTS connectionless-based radio interface to conduct a telephone call or a real time video transmission, we are typically faced either with a relatively large number of retransmissions or a choice of strong channel coding. The former is contradictory to the requirement of real time and the latter lowers the throughput of actual data to an unacceptable level for the transmission of speech or images with a reasonable quality. The relatively short interleaving length makes the situation even worse, because it weakens the performance of the radio interface against bursty transmission errors.
An obvious solution for enhancing the applicability of packet data radio interfaces to the transmission of real time audio and/or video would be to specify at least one specific bearer type for them. Examples of channel coding and interleaving optimized for the transmission of real time audio and/or video are abundant in the field of digital mobile telephony, so the person skilled in the art would have no difficulties in specifying suitable characteristics for a “speech bearer” of the like in association with e.g. GPRS. However, it is not obvious how and when should it be decided to allocate such a bearer to a certain connection that is to be set up.